Abstract: An Nth order linear differential microphone array (LDMA) with a steerable beamformer may be constructed by specifying a target beampattern for the LDMA at a steering angle ?. An Nth order polynomial associated with the target beampattern may then be generated. A relationship between the nulls of the polynomial and the steering angle ? is determined and then a value of one of the nulls is determined based on N?1 assigned values for the other nulls and the determined relationship between the nulls of the polynomial and the steering angle ?. The steerable beamformer may be generated based on the determined null value and the N?1 assigned null values. The N?1 assigned null values may be associated with the N?1 nulls of the polynomial that are of less than Nth order and the determined null value may be associated with the null of the polynomial that is of Nth order.

Abstract: First and second computer systems and respective first and second microphones thereof receive respective portions of a same audio input signal. Audio buffers received respectively from the first and second computer system include data encoded from respective microphone inputs of the first and the second computer systems. The received audio buffers are synchronized and corrected for gain differences between the received audio buffers to produce corrected audio buffers. The corrected audio buffers are mixed into an output buffer. The synchronization reduces echo when the output buffer is played at a remote peer computer system.

Abstract: The present invention relates to a microphone array based pickup method, comprising: performing voice activation detection using one channel voice signal among multichannel voice signals picked up and output by a microphone array, and determining if a voice activation signal occurs; locating the voice source by using the multichannel voice signals output by the microphone array to obtain the voice source locating direction; enhancing a voice signal in the voice source locating direction to obtain an enhanced voice signal; conducting voice wakeup detection on the enhanced voice signal and determining if a voice wakeup is detected; picking up and outputting the multichannel voice signals by the microphone array; Step 6: processing the multichannel voice signals picked up by the microphone array into one channel enhanced voice, and outputting the one channel enhanced voice as a finally picked up voice.

Abstract: An acoustic delay measurement apparatus measures an audio delay introduced by an audio system. A programmable delay buffer, which introduces a programmable delay into the audio stream, receives an audio stream from an audio source and outputs a reference signal representing the audio stream. An adaptive filter is responsive to the reference signal to generate an estimate signal to match on convergence of the adaptive filter an audio signal output by the audio system, which is the audio stream delayed by an amount representative of the audio delay introduced by the audio system. A processor including a coefficient analysis block reads coefficients in the adaptive filter after convergence, computes a delay introduced into said estimate signal by the adaptive filter, and adds the computed delay to the programmable delay buffer to provide a measurement of the audio delay.

Abstract: Disclosed is a method for delaying audio from a received signal prior to processing in an acoustic echo cancellation device, comprising: receiving an audio signal at an audio extraction and delay device; determining a first amount of delay to apply to the received audio signal such that acoustic echo cancellation (AEC) processing can occur; applying the first amount of delay to the received audio signal; transmitting the un-delayed audio signal to an AEC circuit; and transmitting the delayed audio signal to one or more speakers.

Abstract: Disclosed is a microphone system, comprising: one or more microphones each of which is adapted to receive acoustic audio signals and convert the same from acoustical energy signals into electrical audio signals and output the same as microphone output audio signals; and an acoustic echo cancellation (AEC) device adapted to receive the microphone output audio signals, and wherein the AEC device also receives as a second input a reference audio signal from a network, and wherein the AEC device is further adapted to cancel substantially all of the reference audio signal from the microphone output audio signals and output the same as a corrected audio signal, and wherein the reference audio signal comprises an audio signal generated by an external audio system.

Abstract: Disclosed is a method for determining one or more spoken words, comprising: receiving acoustic audio signals at one or more microphones within a microphone system, and converting the same from acoustical energy signals into electrical audio signals and outputting them as microphone output audio signals; receiving the microphone output audio signals from the microphone device at a first input of an acoustic echo cancellation (AEC) device, and receiving a reference input signal at a second input of the AEC device; cancelling substantially all of the reference audio signal from the microphone output audio signal; and outputting the same as a corrected audio signal, and wherein the reference audio signal comprises an audio signal generated by an external audio system.

Abstract: Various aspects include approaches for intelligent acoustic beam steering. In particular aspects, a computer-implemented method of controlling a microphone array includes: generating an acoustic map including an acoustic description of a physical environment proximate the speaker system; focusing the microphone array in a direction based upon the acoustic map prior to receiving a voice command input at the microphone array, where generating the acoustic map includes: detecting a location of at least one noise source in the physical environment; and creating a microphone array filter configured to at least partially reject the at least one noise source; receiving voice feedback input from a user at the microphone array; and updating the acoustic map of the physical environment based upon the received voice feedback input.

Abstract: One embodiment id directed towards a noise reduction system that includes a detector block that is configured to detect noise components in an input signal based on a signal-to-noise ratio spectrum of the input signal. The noise reduction system also includes a masking block operatively coupled with the detector block and configured to generate a final spectral noise removal mask and to apply the final spectral noise removal mask to the input signal if noise components in the input signal are detected, the final spectral noise removal mask being configured to suppress the noise components in the input signal, when applied.

Abstract: In accordance with embodiments of the present disclosure, a digital microphone system may include a microphone transducer and a digital processing system. The microphone transducer may be configured to generate an analog input signal indicative of audio sounds incident upon the microphone transducer. The digital processing system may be configured to convert the analog input signal into a first digital signal having a plurality (e.g., more than 3) of quantization levels, and in the digital domain, process the first digital signal to compress the first digital signal into a second digital signal having fewer quantization levels (e.g., +1, 0, ?1) than that of the first digital signal.

Abstract: Disclosed are a sound sharing apparatus and a sound sharing method. The sound sharing apparatus according to one embodiment of the present disclosure includes at least one processor configured to implement: a modifier configured to change a default audio render driver of a local machine from a first audio render driver to a second audio render driver; a capturer configured to capture audio data transmitted to the second audio render driver; and a mixer configured to mix the captured audio data: i) with first voice data to output first mixed data, wherein the first voice data is received from a remote machine connected to the local machine through a network, or ii) with second voice data to output second mixed data, wherein the second voice data is received input through a microphone of the local machine.

Abstract: A method including: receiving multiple pieces of notice information; receiving detection information indicating whether or not the user is present around a speaker, notifying the user of content of first notice information in the multiple pieces of notice information by using the speaker, in a case that it is determined based on the detection information that the user is present around the speaker, and notifying the user of content of second notice information different from the first notice information, by using the speaker in a case that it is determined that the user is present around the speaker when the notification of the first notice information by using the speaker is complete.

Abstract: A system may include a server configured to detect speech data from media content and to divide the detected speech data into one or more speech data segments in accordance with at least a respective speaker and a break in the detected speech data; and a media content playing device configured to receive the speech data segments from the server, to receive, from an input device, a control signal to play the media content, and to skip forward or rewind to play the media content starting at the identified starting point corresponding to a first one of the respective speech data segments.

Abstract: An echo cancellation system that synchronizes output audio data with input audio data in a heterogeneous system. The system may increment a counter as outgoing audio frames are sent to a digital-to-analog converter in a speaker. As incoming audio frames are received by an analog-to-digital converter in a microphone, the system may copy contents of the counter into the incoming audio frames. Based on the contents of the counter, the incoming audio frames may be associated with corresponding outgoing audio frames. After synchronizing the incoming audio frames and the outgoing audio frames, the system may perform Acoustic Echo Cancellation by removing the outgoing audio frames from the incoming audio frames.

Abstract: Determining the end of an utterance for purposes of automatic speech recognition (ASR) may be improved with a system that provides early results and/or incorporates semantic tagging. Early ASR results of an incoming utterance may be prepared based at least in part on an estimated endpoint and processed by a natural language understanding (NLU) process while final results, based at least in part on a final endpoint, are determined. If the early results match the final results, the early NLU results are already prepared for early execution. The endpoint may also be determined based at least in part on the content of the utterance, as represented by semantic tagging output from ASR processing. If the tagging indicate completion of a logical statement, an endpoint may be declared, or a threshold for silent frames prior to declaring an endpoint may be adjusted.

Abstract: A computer-implemented system and method for distributing messages by discussion group is provided. Discussion groups are monitored. An identity of a participant in each of two or more of the discussion groups is verified. Voice signals are received from the participant and speech from the participant is distinguished from ambient noise and third-party conversations in the voice signals. One of the discussion groups, specified by the participant to be associated with the voice stream, is identified. The speech is transmitted as a voice message to other participants in the identified discussion group.

Abstract: System and methods are provided for voice enhancement in audio conferencing among a plurality of participants. An example system includes a signal processor, a pre-processing component, and a voice-enhancement component. The signal processor is configured to generate a first mixed signal based at least in part on a first audio signal associated with a first remote participant and a local audio signal associated with a local participant. The pre-processing component is configured to generate a first input signal and a second input signal based at least in part on the first mixed signal and a second audio signal associated with a second remote participant. In addition, the voice-enhancement component is configured to generate a first output signal to be transmitted to the second remote participant based at least in part on the first input signal and the second input signal.

Abstract: Apparatus, computer-readable storage medium, and method associated with speech recognition are described. In embodiments, a mobile phone may include a processor; and a speech recognition module coupled with the processor. The voice recognition module may be configured to recognize one or more voice commands and may include first echo cancellation logic and second echo cancellation logic to be selectively employed during recognition of voice commands. Employment of the first and second echo cancellation logic respectively may cause the mobile phone to variably consume a first and second amount of energy, with the second amount of energy being less than the first amount energy.

Abstract: Methods and apparatus for determining the quality of a communication link transmitting a specified packet type are disclosed. The methodology includes configuring first packets, which share transmission characteristics with a specific packet type, such as a Voice over IP packet, such that the first packets experience similar communication link treatment as the specific packet type. The first packets are then transmitted over the communication link to a quality monitoring server located near a termination equipment of the particular communication link. Second packets, which are transmitted by the server in response to the first packets, are received and evaluated to determine the quality of the communication link based on characteristics of the second packets. Corresponding apparatus are also disclosed.

Abstract: According to one embodiment, a first processing module adds, to a first queue, output sound data output from a first task, with a time stamp attached thereto. A second processing module adds, to a second queue, input sound data received from a microphone, with a time stamp attached thereto. A controller fetches first output sound data as reference data from the first queue, the first output sound data having a time stamp whose time difference from a time stamp of first input sound data in the second queue falls within a predetermined range. An echo canceller performs echo cancelling processing to cancel an echo component in the first input sound data based on the reference data.

Abstract: The present invention provides a method and an apparatus for training a Vector coefficient of a vectored digital subscriber line, where the method includes: after a handshake phase is executed for a joining line, calculating, by using a pilot estimation method, an initial precoding coefficient and an initial cancellation coefficient that represent interference from the joining line to a normal working line; and after a channel discovery phase is executed for the joining line, compensating the initial precoding coefficient and the initial cancellation coefficient by using a compensation factor, to obtain a precoding coefficient and a cancellation coefficient that represent the interference from the joining line to the normal working line, wherein the compensation factor is obtained according to a signal power change or a signal phase change of the joining line. The present invention improves efficiency of Vector coefficient training.

Abstract: A new audio echo cancellation (AEC) approach is disclosed. To facilitate echo cancellation, the method adjusts for errors (called drift) in sampling rates for both capturing audio and playing audio. This ensures that the AEC module receives both the signals at precisely the same sampling frequency. Furthermore, the far-end signal and near-end mixed signal are time aligned to ensure that the alignment is suitable for application of AEC techniques. An additional enhancement to reduce errors utilizes a concept of native frequency. A by-product of drift compensation allows for excellent buffer control for capture/playback and buffer overflow/underflow errors from drift errors are eliminated.

Abstract: Embodiments of methods and devices for classifying background noise contained in an audio signal are disclosed. In one embodiment, the device includes a module for extracting from the audio signal a background noise signal, termed the noise signal. Also included is a second that calculates a first parameter, termed the temporal indicator. The temporal indicator relates to the temporal evolution of the noise signal. The second module also calculates a second parameter, termed the frequency indicator. The frequency indicator relates to the frequency spectrum of the noise signal. Finally, the device includes a third module that classifies the background noise by selecting, as a function of the calculated values of the temporal indicator and of the frequency indicator, a class of background noise from among a predefined set of classes of background noise.

Abstract: A plurality of microphones of a communication device is grouped into multiple microphone groups, such that each microphone group includes two or more microphones. For each microphone group, output of the corresponding microphones is processed to form an acoustic null in a corresponding spatial direction, such that sound from the corresponding spatial direction is attenuated in the processed output. One of the microphone groups is selected based on various factors leading to maximal echo attenuation and rejection of reverberant components of the room. The selected microphone group is then used to detect sound from a near end talker of the communication device.

Abstract: System and methods provide acoustic echo monitoring and cancellation for real time media processing in an internet protocol (IP) media server in an IP network. An echo monitor is configured to selectively compare audio streams into and out of the IP media server through a selected port. The comparison determines an occurrence of an echo. An echo canceller in communication with the echo monitor is configured to respond to the determination by the echo monitor so as to remove the echo from at least one of the audio streams. A talk burst detector may be used to detect speech in at least one of the audio streams through the selected port. The echo monitor selectively compares the audio streams in response to a signal from the talk burst detector that indicates detection of speech.

Abstract: Techniques for utilizing blind source separation as a front-end to an acoustic echo canceller are described herein. The techniques include removing a first portion of an acoustic echo from an audio signal using blind source separation and a reference signal. The techniques then further remove a second portion of the acoustic echo using an acoustic echo canceller and the reference signal. Further, output of the blind source separation may be used to improve double-talk detection.

Abstract: Method, user terminal and computer program product for controlling audio signals at the user device during a communication session between the user device and a remote node, in which a primary audio signal is received at audio input means of the user device for transmission to the remote node in the communication session. It is determined whether the user device is operating in (i) a first mode in which secondary audio signals output from the user device are likely to disturb the primary audio signal received at the audio input means, or (ii) a second mode in which secondary audio signals output from the user device are not likely to disturb the primary audio signal received at the audio input means.

Abstract: Systems and methods for adaptive power control are provided. In exemplary embodiments, a primary signal is received. A noise power level of the primary signal is then estimated. The noise power level may then be compared to at least one power threshold. Subsequently, a large power consuming system is controlled based on the comparison of the noise power level to the power threshold.

Abstract: The present disclosure relates to systems and methods for noise cancellation of phone conversation. Some of the systems can include a phone having a microphone and a noise cancellation device. In some examples, the noise cancellation device has a microphone for receiving an input sound wave and positioned proximate the phone microphone, a processor for generating a noise cancellation sound wave based on the input sound wave, and a speaker for outputting the noise cancellation sound wave. The phone microphone and the noise cancellation device microphone can be shared or different microphones, and can also be microphone is integral to the phone. The shared or separate microphones can be part of a phone headset.

Abstract: A noise canceling headphone includes a microphone provided in a headphone housing, collecting external sound, a canceling signal generating circuit generating a canceling signal in response to a signal output from the microphone which cancels external sound passing through the housing without being attenuated by a sound insulating characteristic of the headphone, a compensating signal generating circuit generating a compensating signal that compensates for the external sound attenuated by the sound insulating characteristic of the headphone, an adding circuit that adds a musical signal input from the exterior to the compensating signal, and a speaker unit that outputs an output signal from the adding circuit and the canceling signal.

Abstract: Noise reduction is provided to audio captured by a headset by employing spatially separated microphones provided by a headset and a mobile phone. Primary audio is captured by a headset microphone and includes both voice audio and ambient noise. Secondary audio is captured by a mobile phone microphone and includes ambient audio. Noise reduction is performed using the primary and secondary audio to generate a noise-reduce audio.

Abstract: A new acoustic echo suppressor and method for acoustic echo suppression is described herein. Exemplary embodiments of the acoustic echo suppressor use one linear regression model for each subband. The linear regression model for each subband may operate on the squared magnitude of the input samples as well as corresponding cross-products. In this way, accurate and robust estimates of the echo signal in each subband can be obtained, thereby providing good echo reduction while keeping the signal distortion low.

Abstract: A method is provided for encoding multiple microphone signals into a composite source-separable audio (SSA) signal, conducive for transmission over a voice network. The embodiments enable the processing of source separation of the target voice signal from its ambient sound to be performed at any point in the voice communication network, including the internet cloud. A multiplicity of processing is possible over the SSA signal, based on the intended voice application. The level of processing is adapted with the availability of the processing power at the chosen processing node in the network in one embodiment. An apparatus for separating out the target source voice from its ambient sound is also provided. The apparatus includes a directed source separation (DSS) unit, which processes the two virtual microphone signals in the SSA representation, to generate a new SSA signal including the enhanced target voice and the enhanced ambient noise.

Abstract: Echo cancellation is handled using a pattern classification technique. During a training phase, the communications device is trained to learn patterns of signal inputs that correspond to certain communication modes. After numerous different patterns have been classified by mode, the classified patterns are used during real-time use of the communications device in order to determine, dynamically, the mode in which the communications device is currently operating. The communications device may then apply, to the microphone-produced signal, a suppression action.

Abstract: A signal processing system enhances communication. When an audio signal is detected, an echo component of the detected audio signal may be estimated. A near party communication device may receive an audio signal from a remote party communication device. A characteristic of the received audio signal may be adjusted based on the echo component of the detected audio signal.

Abstract: A call directing system receives an incoming call from a caller. The caller is prompted to speak, thus enabling a prosody analyzer to generate an analysis of a prosody of the caller's voice. This analysis provides a basis for generating a caller profile that describes caller preferences of the caller. Based on the analysis of the prosody of the caller's voice and the generated caller profile, the call is directed to a particular call recipient.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Abstract: A method of enhancing an audio signal includes the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-

Abstract: Method, user terminal and computer program product for controlling audio signals at the user device during a communication session between the user device and a remote node, in which a primary audio signal is received at audio input means of the user device for transmission to the remote node in the communication session. It is determined whether the user device is operating in (i) a first mode in which secondary audio signals output from the user device are likely to disturb the primary audio signal received at the audio input means, or (ii) a second mode in which secondary audio signals output from the user device are not likely to disturb the primary audio signal received at the audio input means.

Abstract: Embodiments of the present disclosure generally include a method and apparatus for minimizing acoustic echo in video conference. In one embodiment, a computer implemented method for minimizing acoustic echo in video conferencing comprises determining position information wherein the position information is derivable from a video stream and processing the position information using a computer to identify at least one direction wherein the one direction minimizes acoustic echo.

Abstract: An echo canceller includes an initial delay estimator producing a reference signal containing an identified frequency component masked by means of frequency masking effect to output the reference signal on an echo path, and detecting, from a sending voice signal, an echo component of the identified frequency component derived from the reference signal. The estimator finds an initial delay amount on the echo path on the basis of a forming timing of the reference signal containing the identified frequency component and a detection timing of the echo component in the identified frequency component in the sending voice signal. The echo canceller includes a delay circuit for delaying a received voice signal by the initial delay amount on the echo path obtained from the estimator to supply a resulting signal to a pseudo echo generator.

Abstract: An exemplary embodiment of the present invention is directed toward a method and system for cancelling line echo in the presence of a known secondary audio signal. Filter adaption is enabled in the presence of a known secondary audio source such as the sound of a computer game, a music signal or other secondary audio sources that would otherwise prevent echo cancellation due to an apparent double talk condition. It is emphasized that this abstract is provided to comply with the rules requiring an abstract which will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or the meaning of the claims.

Abstract: A regression-based residual echo suppression (RES) system and process for suppressing the portion of the microphone signal corresponding to a playback of a speaker audio signal that was not suppressed by an acoustic echo canceller (AEC). In general, a prescribed regression technique is used between a prescribed spectral attribute of multiple past and present, fixed-length, periods (e.g., frames) of the speaker signal and the same spectral attribute of a current period (e.g., frame) of the echo residual in the output of the AEC. This automatically takes into consideration the correlation between the time periods of the speaker signal. The parameters of the regression can be easily tracked using adaptive methods. Multiple applications of RES can be used to produce better results and this system and process can be applied to stereo-RES as well.

Abstract: A mobile communications device contains at least two microphones. One microphone is designated by a selector to provide a voice dominant signal and another microphone is designated to provide a noise or echo dominant signal, for a call or a recording. The selector communicates the designations to a switch that routes the selected microphone signals to the inputs of a processor for voice signal enhancement. The selected voice dominant signal is then enhanced by suppressing ambient noise or canceling echo therein, based on the selected noise or echo dominant signal. The designation of microphones may change at any instant during the call or recording depending on various factors, e.g. based on the quality of the microphone signals. Other embodiments are also described.

Abstract: A combined sidetone and hybrid balance apparatus and method of operating same are disclosed. An embodiment of the present invention may provide both a hybrid balance mode of operation and a sidetone generation mode of operation within a single integrated circuit device. The functionality provided by an embodiment of the present invention may be used in both Internet protocol (IP)-based telephones and residential gateways, and may be incorporated within the IP telephone chip used in such devices. An embodiment of the present invention may be used to reduce processor demand and the cost of materials, permitting greater IP telephone or residential gateway functionality at lower cost.

Abstract: A speech enhancement device and a method for the same are included. The device includes a down-converter, a speech enhancement processor, and an up-converter. The method includes steps of down-converting audio signals to generate down-converted audio signals; performing speech enhancement on the down-converted audio signals to generate speech-enhanced audio signals; and up-converting the speech enhancement audio signals to generate up-converted audio signals.

Abstract: According to one embodiment, in response to a first acoustic signal output, a second acoustic signal is input. A filter unit is configured to generate a third acoustic signal by convoluting the first acoustic signal with coefficients. A subtraction unit is configured to generate a fourth acoustic signal by subtracting the third acoustic signal from the second acoustic signal. An estimation unit is configured to decide whether a sound volume of the first acoustic signal is below a predetermined threshold, and to set a sound volume of the second acoustic signal as a non-echo sound level when the sound volume of the first acoustic signal is below the predetermined threshold. A determination unit is configured to determine a step size to correct the coefficients using the non-echo sound level. A correction unit is configured to correct the coefficients using the step size.

Abstract: A method and apparatus for cross-talk resistant adaptive noise cancellation.

Abstract: In an echo suppression apparatus, an adaptive filter estimates an echo path of a near end, and generates a pseudo echo signal of a reception signal received from a far end. A subtractor subtracts the pseudo echo signal from a near-end signal including an echo signal of the reception signal, a near-end sound and a background noise, thereby generating an echo-canceled signal. A background noise estimation unit estimates a spectrum of the background noise. A non-linear processing unit performs spectrum subtraction of the estimated spectrum of the background noise from the spectrum of the echo-canceled signal, and controls a gain of the spectrum of the echo-canceled signal in response to the result of the spectrum subtraction, thereby obtaining a spectrum of a transmission signal transmitted to the far end. A threshold calculation unit calculates a threshold value used to determine presence or absence of the residual echo in the echo-canceled signal.

Abstract: In an echo canceller, a specific frequency component eliminator eliminates a specific frequency component of a specific frequency from a receiver signal to output a resultant signal to an echo path. A specific frequency component detector detects a frequency component of the same frequency as the specific frequency eliminated by the specific frequency component eliminator from the transmitter signal. A noise calculator finds noise power on the basis of power of the specific frequency component detected by the specific frequency component detector, and finds total power including noise and an echo component on the basis of power of a frequency component including the echo component. A control parameter calculator uses the noise power and the total power found by the noise calculator to find a control parameter of the echo canceller.